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ffmpeg/libavcodec/dcadec.c

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/*
* DCA compatible decoder
* Copyright (C) 2004 Gildas Bazin
* Copyright (C) 2004 Benjamin Zores
* Copyright (C) 2006 Benjamin Larsson
* Copyright (C) 2007 Konstantin Shishkov
* Copyright (C) 2012 Paul B Mahol
* Copyright (C) 2014 Niels Möller
*
* This file is part of Libav.
*
* Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <math.h>
#include <stddef.h>
#include <stdio.h>
#include "libavutil/attributes.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "libavutil/internal.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/mathematics.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "avcodec.h"
#include "dca.h"
#include "dca_syncwords.h"
#include "dcadata.h"
#include "dcadsp.h"
#include "dcahuff.h"
#include "fft.h"
#include "fmtconvert.h"
#include "get_bits.h"
#include "internal.h"
#include "mathops.h"
#include "profiles.h"
#include "put_bits.h"
#include "synth_filter.h"
#if ARCH_ARM
# include "arm/dca.h"
#endif
enum DCAMode {
DCA_MONO = 0,
DCA_CHANNEL,
DCA_STEREO,
DCA_STEREO_SUMDIFF,
DCA_STEREO_TOTAL,
DCA_3F,
DCA_2F1R,
DCA_3F1R,
DCA_2F2R,
DCA_3F2R,
DCA_4F2R
};
/* -1 are reserved or unknown */
static const int dca_ext_audio_descr_mask[] = {
DCA_EXT_XCH,
-1,
DCA_EXT_X96,
DCA_EXT_XCH | DCA_EXT_X96,
-1,
-1,
DCA_EXT_XXCH,
-1,
};
/* Tables for mapping dts channel configurations to libavcodec multichannel api.
* Some compromises have been made for special configurations. Most configurations
* are never used so complete accuracy is not needed.
*
* L = left, R = right, C = center, S = surround, F = front, R = rear, T = total, OV = overhead.
* S -> side, when both rear and back are configured move one of them to the side channel
* OV -> center back
* All 2 channel configurations -> AV_CH_LAYOUT_STEREO
*/
static const uint64_t dca_core_channel_layout[] = {
AV_CH_FRONT_CENTER, ///< 1, A
AV_CH_LAYOUT_STEREO, ///< 2, A + B (dual mono)
AV_CH_LAYOUT_STEREO, ///< 2, L + R (stereo)
AV_CH_LAYOUT_STEREO, ///< 2, (L + R) + (L - R) (sum-difference)
AV_CH_LAYOUT_STEREO, ///< 2, LT + RT (left and right total)
AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER, ///< 3, C + L + R
AV_CH_LAYOUT_STEREO | AV_CH_BACK_CENTER, ///< 3, L + R + S
AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER, ///< 4, C + L + R + S
AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT, ///< 4, L + R + SL + SR
AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_SIDE_LEFT |
AV_CH_SIDE_RIGHT, ///< 5, C + L + R + SL + SR
AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |
AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER, ///< 6, CL + CR + L + R + SL + SR
AV_CH_LAYOUT_STEREO | AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT |
AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER, ///< 6, C + L + R + LR + RR + OV
AV_CH_FRONT_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |
AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_BACK_CENTER |
AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT, ///< 6, CF + CR + LF + RF + LR + RR
AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER |
AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |
AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT, ///< 7, CL + C + CR + L + R + SL + SR
AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |
AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |
AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT, ///< 8, CL + CR + L + R + SL1 + SL2 + SR1 + SR2
AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER |
AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |
AV_CH_SIDE_LEFT | AV_CH_BACK_CENTER | AV_CH_SIDE_RIGHT, ///< 8, CL + C + CR + L + R + SL + S + SR
};
#define DCA_DOLBY 101 /* FIXME */
#define DCA_CHANNEL_BITS 6
#define DCA_CHANNEL_MASK 0x3F
#define DCA_LFE 0x80
#define HEADER_SIZE 14
#define DCA_NSYNCAUX 0x9A1105A0
/** Bit allocation */
typedef struct BitAlloc {
int offset; ///< code values offset
int maxbits[8]; ///< max bits in VLC
int wrap; ///< wrap for get_vlc2()
VLC vlc[8]; ///< actual codes
} BitAlloc;
static BitAlloc dca_bitalloc_index; ///< indexes for samples VLC select
static BitAlloc dca_tmode; ///< transition mode VLCs
static BitAlloc dca_scalefactor; ///< scalefactor VLCs
static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs
static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba,
int idx)
{
return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) +
ba->offset;
}
static av_cold void dca_init_vlcs(void)
{
static int vlcs_initialized = 0;
int i, j, c = 14;
static VLC_TYPE dca_table[23622][2];
if (vlcs_initialized)
return;
dca_bitalloc_index.offset = 1;
dca_bitalloc_index.wrap = 2;
for (i = 0; i < 5; i++) {
dca_bitalloc_index.vlc[i].table = &dca_table[ff_dca_vlc_offs[i]];
dca_bitalloc_index.vlc[i].table_allocated = ff_dca_vlc_offs[i + 1] - ff_dca_vlc_offs[i];
init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12,
bitalloc_12_bits[i], 1, 1,
bitalloc_12_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
}
dca_scalefactor.offset = -64;
dca_scalefactor.wrap = 2;
for (i = 0; i < 5; i++) {
dca_scalefactor.vlc[i].table = &dca_table[ff_dca_vlc_offs[i + 5]];
dca_scalefactor.vlc[i].table_allocated = ff_dca_vlc_offs[i + 6] - ff_dca_vlc_offs[i + 5];
init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,
scales_bits[i], 1, 1,
scales_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
}
dca_tmode.offset = 0;
dca_tmode.wrap = 1;
for (i = 0; i < 4; i++) {
dca_tmode.vlc[i].table = &dca_table[ff_dca_vlc_offs[i + 10]];
dca_tmode.vlc[i].table_allocated = ff_dca_vlc_offs[i + 11] - ff_dca_vlc_offs[i + 10];
init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
tmode_bits[i], 1, 1,
tmode_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
}
for (i = 0; i < 10; i++)
for (j = 0; j < 7; j++) {
if (!bitalloc_codes[i][j])
break;
dca_smpl_bitalloc[i + 1].offset = bitalloc_offsets[i];
dca_smpl_bitalloc[i + 1].wrap = 1 + (j > 4);
dca_smpl_bitalloc[i + 1].vlc[j].table = &dca_table[ff_dca_vlc_offs[c]];
dca_smpl_bitalloc[i + 1].vlc[j].table_allocated = ff_dca_vlc_offs[c + 1] - ff_dca_vlc_offs[c];
init_vlc(&dca_smpl_bitalloc[i + 1].vlc[j], bitalloc_maxbits[i][j],
bitalloc_sizes[i],
bitalloc_bits[i][j], 1, 1,
bitalloc_codes[i][j], 2, 2, INIT_VLC_USE_NEW_STATIC);
c++;
}
vlcs_initialized = 1;
}
static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
{
while (len--)
*dst++ = get_bits(gb, bits);
}
static int dca_parse_audio_coding_header(DCAContext *s, int base_channel)
{
int i, j;
static const uint8_t adj_table[4] = { 16, 18, 20, 23 };
static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
s->audio_header.total_channels = get_bits(&s->gb, 3) + 1 + base_channel;
s->audio_header.prim_channels = s->audio_header.total_channels;
if (s->audio_header.prim_channels > DCA_PRIM_CHANNELS_MAX)
s->audio_header.prim_channels = DCA_PRIM_CHANNELS_MAX;
for (i = base_channel; i < s->audio_header.prim_channels; i++) {
s->audio_header.subband_activity[i] = get_bits(&s->gb, 5) + 2;
if (s->audio_header.subband_activity[i] > DCA_SUBBANDS)
s->audio_header.subband_activity[i] = DCA_SUBBANDS;
}
for (i = base_channel; i < s->audio_header.prim_channels; i++) {
s->audio_header.vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
if (s->audio_header.vq_start_subband[i] > DCA_SUBBANDS)
s->audio_header.vq_start_subband[i] = DCA_SUBBANDS;
}
get_array(&s->gb, s->audio_header.joint_intensity + base_channel,
s->audio_header.prim_channels - base_channel, 3);
get_array(&s->gb, s->audio_header.transient_huffman + base_channel,
s->audio_header.prim_channels - base_channel, 2);
get_array(&s->gb, s->audio_header.scalefactor_huffman + base_channel,
s->audio_header.prim_channels - base_channel, 3);
get_array(&s->gb, s->audio_header.bitalloc_huffman + base_channel,
s->audio_header.prim_channels - base_channel, 3);
/* Get codebooks quantization indexes */
if (!base_channel)
memset(s->audio_header.quant_index_huffman, 0, sizeof(s->audio_header.quant_index_huffman));
for (j = 1; j < 11; j++)
for (i = base_channel; i < s->audio_header.prim_channels; i++)
s->audio_header.quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
/* Get scale factor adjustment */
for (j = 0; j < 11; j++)
for (i = base_channel; i < s->audio_header.prim_channels; i++)
s->audio_header.scalefactor_adj[i][j] = 16;
for (j = 1; j < 11; j++)
for (i = base_channel; i < s->audio_header.prim_channels; i++)
if (s->audio_header.quant_index_huffman[i][j] < thr[j])
s->audio_header.scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
if (s->crc_present) {
/* Audio header CRC check */
get_bits(&s->gb, 16);
}
s->current_subframe = 0;
s->current_subsubframe = 0;
return 0;
}
static int dca_parse_frame_header(DCAContext *s)
{
init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
/* Sync code */
skip_bits_long(&s->gb, 32);
/* Frame header */
s->frame_type = get_bits(&s->gb, 1);
s->samples_deficit = get_bits(&s->gb, 5) + 1;
s->crc_present = get_bits(&s->gb, 1);
s->sample_blocks = get_bits(&s->gb, 7) + 1;
s->frame_size = get_bits(&s->gb, 14) + 1;
if (s->frame_size < 95)
return AVERROR_INVALIDDATA;
s->amode = get_bits(&s->gb, 6);
s->sample_rate = avpriv_dca_sample_rates[get_bits(&s->gb, 4)];
if (!s->sample_rate)
return AVERROR_INVALIDDATA;
s->bit_rate_index = get_bits(&s->gb, 5);
s->bit_rate = ff_dca_bit_rates[s->bit_rate_index];
if (!s->bit_rate)
return AVERROR_INVALIDDATA;
skip_bits1(&s->gb); // always 0 (reserved, cf. ETSI TS 102 114 V1.4.1)
s->dynrange = get_bits(&s->gb, 1);
s->timestamp = get_bits(&s->gb, 1);
s->aux_data = get_bits(&s->gb, 1);
s->hdcd = get_bits(&s->gb, 1);
s->ext_descr = get_bits(&s->gb, 3);
s->ext_coding = get_bits(&s->gb, 1);
s->aspf = get_bits(&s->gb, 1);
s->lfe = get_bits(&s->gb, 2);
s->predictor_history = get_bits(&s->gb, 1);
if (s->lfe > 2) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE value: %d\n", s->lfe);
return AVERROR_INVALIDDATA;
}
/* TODO: check CRC */
if (s->crc_present)
s->header_crc = get_bits(&s->gb, 16);
s->multirate_inter = get_bits(&s->gb, 1);
s->version = get_bits(&s->gb, 4);
s->copy_history = get_bits(&s->gb, 2);
s->source_pcm_res = get_bits(&s->gb, 3);
s->front_sum = get_bits(&s->gb, 1);
s->surround_sum = get_bits(&s->gb, 1);
s->dialog_norm = get_bits(&s->gb, 4);
/* FIXME: channels mixing levels */
s->output = s->amode;
if (s->lfe)
s->output |= DCA_LFE;
/* Primary audio coding header */
s->audio_header.subframes = get_bits(&s->gb, 4) + 1;
return dca_parse_audio_coding_header(s, 0);
}
static inline int get_scale(GetBitContext *gb, int level, int value, int log2range)
{
if (level < 5) {
/* huffman encoded */
value += get_bitalloc(gb, &dca_scalefactor, level);
value = av_clip(value, 0, (1 << log2range) - 1);
} else if (level < 8) {
if (level + 1 > log2range) {
skip_bits(gb, level + 1 - log2range);
value = get_bits(gb, log2range);
} else {
value = get_bits(gb, level + 1);
}
}
return value;
}
static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
{
/* Primary audio coding side information */
int j, k;
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
if (!base_channel) {
s->subsubframes[s->current_subframe] = get_bits(&s->gb, 2) + 1;
s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3);
}
for (j = base_channel; j < s->audio_header.prim_channels; j++) {
for (k = 0; k < s->audio_header.subband_activity[j]; k++)
s->dca_chan[j].prediction_mode[k] = get_bits(&s->gb, 1);
}
/* Get prediction codebook */
for (j = base_channel; j < s->audio_header.prim_channels; j++) {
for (k = 0; k < s->audio_header.subband_activity[j]; k++) {
if (s->dca_chan[j].prediction_mode[k] > 0) {
/* (Prediction coefficient VQ address) */
s->dca_chan[j].prediction_vq[k] = get_bits(&s->gb, 12);
}
}
}
/* Bit allocation index */
for (j = base_channel; j < s->audio_header.prim_channels; j++) {
for (k = 0; k < s->audio_header.vq_start_subband[j]; k++) {
if (s->audio_header.bitalloc_huffman[j] == 6)
s->dca_chan[j].bitalloc[k] = get_bits(&s->gb, 5);
else if (s->audio_header.bitalloc_huffman[j] == 5)
s->dca_chan[j].bitalloc[k] = get_bits(&s->gb, 4);
else if (s->audio_header.bitalloc_huffman[j] == 7) {
av_log(s->avctx, AV_LOG_ERROR,
"Invalid bit allocation index\n");
return AVERROR_INVALIDDATA;
} else {
s->dca_chan[j].bitalloc[k] =
get_bitalloc(&s->gb, &dca_bitalloc_index, s->audio_header.bitalloc_huffman[j]);
}
if (s->dca_chan[j].bitalloc[k] > 26) {
ff_dlog(s->avctx, "bitalloc index [%i][%i] too big (%i)\n",
j, k, s->dca_chan[j].bitalloc[k]);
return AVERROR_INVALIDDATA;
}
}
}
/* Transition mode */
for (j = base_channel; j < s->audio_header.prim_channels; j++) {
for (k = 0; k < s->audio_header.subband_activity[j]; k++) {
s->dca_chan[j].transition_mode[k] = 0;
if (s->subsubframes[s->current_subframe] > 1 &&
k < s->audio_header.vq_start_subband[j] && s->dca_chan[j].bitalloc[k] > 0) {
s->dca_chan[j].transition_mode[k] =
get_bitalloc(&s->gb, &dca_tmode, s->audio_header.transient_huffman[j]);
}
}
}
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
for (j = base_channel; j < s->audio_header.prim_channels; j++) {
const uint32_t *scale_table;
int scale_sum, log_size;
memset(s->dca_chan[j].scale_factor, 0,
s->audio_header.subband_activity[j] * sizeof(s->dca_chan[j].scale_factor[0][0]) * 2);
if (s->audio_header.scalefactor_huffman[j] == 6) {
scale_table = ff_dca_scale_factor_quant7;
log_size = 7;
} else {
scale_table = ff_dca_scale_factor_quant6;
log_size = 6;
}
/* When huffman coded, only the difference is encoded */
scale_sum = 0;
for (k = 0; k < s->audio_header.subband_activity[j]; k++) {
if (k >= s->audio_header.vq_start_subband[j] || s->dca_chan[j].bitalloc[k] > 0) {
scale_sum = get_scale(&s->gb, s->audio_header.scalefactor_huffman[j], scale_sum, log_size);
s->dca_chan[j].scale_factor[k][0] = scale_table[scale_sum];
}
if (k < s->audio_header.vq_start_subband[j] && s->dca_chan[j].transition_mode[k]) {
/* Get second scale factor */
scale_sum = get_scale(&s->gb, s->audio_header.scalefactor_huffman[j], scale_sum, log_size);
s->dca_chan[j].scale_factor[k][1] = scale_table[scale_sum];
}
}
}
/* Joint subband scale factor codebook select */
for (j = base_channel; j < s->audio_header.prim_channels; j++) {
/* Transmitted only if joint subband coding enabled */
if (s->audio_header.joint_intensity[j] > 0)
s->dca_chan[j].joint_huff = get_bits(&s->gb, 3);
}
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
/* Scale factors for joint subband coding */
for (j = base_channel; j < s->audio_header.prim_channels; j++) {
int source_channel;
/* Transmitted only if joint subband coding enabled */
if (s->audio_header.joint_intensity[j] > 0) {
int scale = 0;
source_channel = s->audio_header.joint_intensity[j] - 1;
/* When huffman coded, only the difference is encoded
* (is this valid as well for joint scales ???) */
for (k = s->audio_header.subband_activity[j];
k < s->audio_header.subband_activity[source_channel]; k++) {
scale = get_scale(&s->gb, s->dca_chan[j].joint_huff, 64 /* bias */, 7);
s->dca_chan[j].joint_scale_factor[k] = scale; /*joint_scale_table[scale]; */
}
if (!(s->debug_flag & 0x02)) {
av_log(s->avctx, AV_LOG_DEBUG,
"Joint stereo coding not supported\n");
s->debug_flag |= 0x02;
}
}
}
/* Dynamic range coefficient */
if (!base_channel && s->dynrange)
s->dynrange_coef = get_bits(&s->gb, 8);
/* Side information CRC check word */
if (s->crc_present) {
get_bits(&s->gb, 16);
}
/*
* Primary audio data arrays
*/
/* VQ encoded high frequency subbands */
for (j = base_channel; j < s->audio_header.prim_channels; j++)
for (k = s->audio_header.vq_start_subband[j]; k < s->audio_header.subband_activity[j]; k++)
/* 1 vector -> 32 samples */
s->dca_chan[j].high_freq_vq[k] = get_bits(&s->gb, 10);
/* Low frequency effect data */
if (!base_channel && s->lfe) {
/* LFE samples */
int lfe_samples = 2 * s->lfe * (4 + block_index);
int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);
float lfe_scale;
for (j = lfe_samples; j < lfe_end_sample; j++) {
/* Signed 8 bits int */
s->lfe_data[j] = get_sbits(&s->gb, 8);
}
/* Scale factor index */
skip_bits(&s->gb, 1);
s->lfe_scale_factor = ff_dca_scale_factor_quant7[get_bits(&s->gb, 7)];
/* Quantization step size * scale factor */
lfe_scale = 0.035 * s->lfe_scale_factor;
for (j = lfe_samples; j < lfe_end_sample; j++)
s->lfe_data[j] *= lfe_scale;
}
return 0;
}
static void qmf_32_subbands(DCAContext *s, int chans,
float samples_in[DCA_SUBBANDS][SAMPLES_PER_SUBBAND], float *samples_out,
float scale)
{
const float *prCoeff;
int sb_act = s->audio_header.subband_activity[chans];
scale *= sqrt(1 / 8.0);
/* Select filter */
if (!s->multirate_inter) /* Non-perfect reconstruction */
prCoeff = ff_dca_fir_32bands_nonperfect;
else /* Perfect reconstruction */
prCoeff = ff_dca_fir_32bands_perfect;
s->dcadsp.qmf_32_subbands(samples_in, sb_act, &s->synth, &s->imdct,
s->dca_chan[chans].subband_fir_hist,
&s->dca_chan[chans].hist_index,
s->dca_chan[chans].subband_fir_noidea, prCoeff,
samples_out, s->raXin, scale);
}
static QMF64_table *qmf64_precompute(void)
{
unsigned i, j;
QMF64_table *table = av_malloc(sizeof(*table));
if (!table)
return NULL;
for (i = 0; i < 32; i++)
for (j = 0; j < 32; j++)
table->dct4_coeff[i][j] = cos((2 * i + 1) * (2 * j + 1) * M_PI / 128);
for (i = 0; i < 32; i++)
for (j = 0; j < 32; j++)
table->dct2_coeff[i][j] = cos((2 * i + 1) * j * M_PI / 64);
/* FIXME: Is the factor 0.125 = 1/8 right? */
for (i = 0; i < 32; i++)
table->rcos[i] = 0.125 / cos((2 * i + 1) * M_PI / 256);
for (i = 0; i < 32; i++)
table->rsin[i] = -0.125 / sin((2 * i + 1) * M_PI / 256);
return table;
}
/* FIXME: Totally unoptimized. Based on the reference code and
* http://multimedia.cx/mirror/dca-transform.pdf, with guessed tweaks
* for doubling the size. */
static void qmf_64_subbands(DCAContext *s, int chans,
float samples_in[DCA_SUBBANDS_X96K][SAMPLES_PER_SUBBAND],
float *samples_out, float scale)
{
float raXin[64];
float A[32], B[32];
float *raX = s->dca_chan[chans].subband_fir_hist;
float *raZ = s->dca_chan[chans].subband_fir_noidea;
unsigned i, j, k, subindex;
for (i = s->audio_header.subband_activity[chans]; i < DCA_SUBBANDS_X96K; i++)
raXin[i] = 0.0;
for (subindex = 0; subindex < SAMPLES_PER_SUBBAND; subindex++) {
for (i = 0; i < s->audio_header.subband_activity[chans]; i++)
raXin[i] = samples_in[i][subindex];
for (k = 0; k < 32; k++) {
A[k] = 0.0;
for (i = 0; i < 32; i++)
A[k] += (raXin[2 * i] + raXin[2 * i + 1]) * s->qmf64_table->dct4_coeff[k][i];
}
for (k = 0; k < 32; k++) {
B[k] = raXin[0] * s->qmf64_table->dct2_coeff[k][0];
for (i = 1; i < 32; i++)
B[k] += (raXin[2 * i] + raXin[2 * i - 1]) * s->qmf64_table->dct2_coeff[k][i];
}
for (k = 0; k < 32; k++) {
raX[k] = s->qmf64_table->rcos[k] * (A[k] + B[k]);
raX[63 - k] = s->qmf64_table->rsin[k] * (A[k] - B[k]);
}
for (i = 0; i < DCA_SUBBANDS_X96K; i++) {
float out = raZ[i];
for (j = 0; j < 1024; j += 128)
out += ff_dca_fir_64bands[j + i] * (raX[j + i] - raX[j + 63 - i]);
*samples_out++ = out * scale;
}
for (i = 0; i < DCA_SUBBANDS_X96K; i++) {
float hist = 0.0;
for (j = 0; j < 1024; j += 128)
hist += ff_dca_fir_64bands[64 + j + i] * (-raX[i + j] - raX[j + 63 - i]);
raZ[i] = hist;
}
/* FIXME: Make buffer circular, to avoid this move. */
memmove(raX + 64, raX, (1024 - 64) * sizeof(*raX));
}
}
static void lfe_interpolation_fir(DCAContext *s, const float *samples_in,
float *samples_out)
{
/* samples_in: An array holding decimated samples.
* Samples in current subframe starts from samples_in[0],
* while samples_in[-1], samples_in[-2], ..., stores samples
* from last subframe as history.
*
* samples_out: An array holding interpolated samples
*/
int idx;
const float *prCoeff;
int deciindex;
/* Select decimation filter */
if (s->lfe == 1) {
idx = 1;
prCoeff = ff_dca_lfe_fir_128;
} else {
idx = 0;
if (s->exss_ext_mask & DCA_EXT_EXSS_XLL)
prCoeff = ff_dca_lfe_xll_fir_64;
else
prCoeff = ff_dca_lfe_fir_64;
}
/* Interpolation */
for (deciindex = 0; deciindex < 2 * s->lfe; deciindex++) {
s->dcadsp.lfe_fir[idx](samples_out, samples_in, prCoeff);
samples_in++;
samples_out += 2 * 32 * (1 + idx);
}
}
/* downmixing routines */
#define MIX_REAR1(samples, s1, rs, coef) \
samples[0][i] += samples[s1][i] * coef[rs][0]; \
samples[1][i] += samples[s1][i] * coef[rs][1];
#define MIX_REAR2(samples, s1, s2, rs, coef) \
samples[0][i] += samples[s1][i] * coef[rs][0] + samples[s2][i] * coef[rs + 1][0]; \
samples[1][i] += samples[s1][i] * coef[rs][1] + samples[s2][i] * coef[rs + 1][1];
#define MIX_FRONT3(samples, coef) \
t = samples[c][i]; \
u = samples[l][i]; \
v = samples[r][i]; \
samples[0][i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \
samples[1][i] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];
#define DOWNMIX_TO_STEREO(op1, op2) \
for (i = 0; i < 256; i++) { \
op1 \
op2 \
}
static void dca_downmix(float **samples, int srcfmt, int lfe_present,
float coef[DCA_PRIM_CHANNELS_MAX + 1][2],
const int8_t *channel_mapping)
{
int c, l, r, sl, sr, s;
int i;
float t, u, v;
switch (srcfmt) {
case DCA_MONO:
case DCA_4F2R:
av_log(NULL, 0, "Not implemented!\n");
break;
case DCA_CHANNEL:
case DCA_STEREO:
case DCA_STEREO_TOTAL:
case DCA_STEREO_SUMDIFF:
break;
case DCA_3F:
c = channel_mapping[0];
l = channel_mapping[1];
r = channel_mapping[2];
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );
break;
case DCA_2F1R:
s = channel_mapping[2];
DOWNMIX_TO_STEREO(MIX_REAR1(samples, s, 2, coef), );
break;
case DCA_3F1R:
c = channel_mapping[0];
l = channel_mapping[1];
r = channel_mapping[2];
s = channel_mapping[3];
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
MIX_REAR1(samples, s, 3, coef));
break;
case DCA_2F2R:
sl = channel_mapping[2];
sr = channel_mapping[3];
DOWNMIX_TO_STEREO(MIX_REAR2(samples, sl, sr, 2, coef), );
break;
case DCA_3F2R:
c = channel_mapping[0];
l = channel_mapping[1];
r = channel_mapping[2];
sl = channel_mapping[3];
sr = channel_mapping[4];
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
MIX_REAR2(samples, sl, sr, 3, coef));
break;
}
if (lfe_present) {
int lf_buf = ff_dca_lfe_index[srcfmt];
int lf_idx = ff_dca_channels[srcfmt];
for (i = 0; i < 256; i++) {
samples[0][i] += samples[lf_buf][i] * coef[lf_idx][0];
samples[1][i] += samples[lf_buf][i] * coef[lf_idx][1];
}
}
}
#ifndef decode_blockcodes
/* Very compact version of the block code decoder that does not use table
* look-up but is slightly slower */
static int decode_blockcode(int code, int levels, int32_t *values)
{
int i;
int offset = (levels - 1) >> 1;
for (i = 0; i < 4; i++) {
int div = FASTDIV(code, levels);
values[i] = code - offset - div * levels;
code = div;
}
return code;
}
static int decode_blockcodes(int code1, int code2, int levels, int32_t *values)
{
return decode_blockcode(code1, levels, values) |
decode_blockcode(code2, levels, values + 4);
}
#endif
static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };
static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
{
int k, l;
int subsubframe = s->current_subsubframe;
const uint32_t *quant_step_table;
/*
* Audio data
*/
/* Select quantization step size table */
if (s->bit_rate_index == 0x1f)
quant_step_table = ff_dca_lossless_quant;
else
quant_step_table = ff_dca_lossy_quant;
for (k = base_channel; k < s->audio_header.prim_channels; k++) {
int32_t (*subband_samples)[8] = s->dca_chan[k].subband_samples[block_index];
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) {
int m;
/* Select the mid-tread linear quantizer */
int abits = s->dca_chan[k].bitalloc[l];
uint32_t quant_step_size = quant_step_table[abits];
/*
* Extract bits from the bit stream
*/
if (!abits)
memset(subband_samples[l], 0, SAMPLES_PER_SUBBAND *
sizeof(subband_samples[l][0]));
else {
uint32_t rscale;
/* Deal with transients */
int sfi = s->dca_chan[k].transition_mode[l] &&
subsubframe >= s->dca_chan[k].transition_mode[l];
/* Determine quantization index code book and its type.
Select quantization index code book */
int sel = s->audio_header.quant_index_huffman[k][abits];
rscale = (s->dca_chan[k].scale_factor[l][sfi] *
s->audio_header.scalefactor_adj[k][sel] + 8) >> 4;
if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {
if (abits <= 7) {
/* Block code */
int block_code1, block_code2, size, levels, err;
size = abits_sizes[abits - 1];
levels = abits_levels[abits - 1];
block_code1 = get_bits(&s->gb, size);
block_code2 = get_bits(&s->gb, size);
err = decode_blockcodes(block_code1, block_code2,
levels, subband_samples[l]);
if (err) {
av_log(s->avctx, AV_LOG_ERROR,
"ERROR: block code look-up failed\n");
return AVERROR_INVALIDDATA;
}
} else {
/* no coding */
for (m = 0; m < SAMPLES_PER_SUBBAND; m++)
subband_samples[l][m] = get_sbits(&s->gb, abits - 3);
}
} else {
/* Huffman coded */
for (m = 0; m < SAMPLES_PER_SUBBAND; m++)
subband_samples[l][m] = get_bitalloc(&s->gb,
&dca_smpl_bitalloc[abits], sel);
}
s->dcadsp.dequantize(subband_samples[l], quant_step_size, rscale);
}
}
for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) {
int m;
/*
* Inverse ADPCM if in prediction mode
*/
if (s->dca_chan[k].prediction_mode[l]) {
int n;
if (s->predictor_history)
subband_samples[l][0] += (ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
(int64_t)s->dca_chan[k].subband_samples_hist[l][3] +
ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][1] *
(int64_t)s->dca_chan[k].subband_samples_hist[l][2] +
ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][2] *
(int64_t)s->dca_chan[k].subband_samples_hist[l][1] +
ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][3] *
(int64_t)s->dca_chan[k].subband_samples_hist[l][0]) +
(1 << 12) >> 13;
for (m = 1; m < SAMPLES_PER_SUBBAND; m++) {
int64_t sum = ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
(int64_t)subband_samples[l][m - 1];
for (n = 2; n <= 4; n++)
if (m >= n)
sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] *
(int64_t)subband_samples[l][m - n];
else if (s->predictor_history)
sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] *
(int64_t)s->dca_chan[k].subband_samples_hist[l][m - n + 4];
subband_samples[l][m] += (int32_t)(sum + (1 << 12) >> 13);
}
}
}
/* Backup predictor history for adpcm */
for (l = 0; l < DCA_SUBBANDS; l++)
AV_COPY128(s->dca_chan[k].subband_samples_hist[l], &subband_samples[l][4]);
/*
* Decode VQ encoded high frequencies
*/
if (s->audio_header.subband_activity[k] > s->audio_header.vq_start_subband[k]) {
if (!s->debug_flag & 0x01) {
av_log(s->avctx, AV_LOG_DEBUG,
"Stream with high frequencies VQ coding\n");
s->debug_flag |= 0x01;
}
s->dcadsp.decode_hf(subband_samples, s->dca_chan[k].high_freq_vq,
ff_dca_high_freq_vq,
subsubframe * SAMPLES_PER_SUBBAND,
s->dca_chan[k].scale_factor,
s->audio_header.vq_start_subband[k],
s->audio_header.subband_activity[k]);
}
}
/* Check for DSYNC after subsubframe */
if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {
if (get_bits(&s->gb, 16) != 0xFFFF) {
av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");
return AVERROR_INVALIDDATA;
}
}
return 0;
}
static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
{
int k;
if (upsample) {
LOCAL_ALIGNED(32, float, samples, [DCA_SUBBANDS_X96K], [SAMPLES_PER_SUBBAND]);
if (!s->qmf64_table) {
s->qmf64_table = qmf64_precompute();
if (!s->qmf64_table)
return AVERROR(ENOMEM);
}
/* 64 subbands QMF */
for (k = 0; k < s->audio_header.prim_channels; k++) {
int32_t (*subband_samples)[SAMPLES_PER_SUBBAND] =
s->dca_chan[k].subband_samples[block_index];
s->fmt_conv.int32_to_float(samples[0], subband_samples[0],
DCA_SUBBANDS_X96K * SAMPLES_PER_SUBBAND);
if (s->channel_order_tab[k] >= 0)
qmf_64_subbands(s, k, samples,
s->samples_chanptr[s->channel_order_tab[k]],
/* Upsampling needs a factor 2 here. */
M_SQRT2 / 32768.0);
}
} else {
/* 32 subbands QMF */
LOCAL_ALIGNED(32, float, samples, [DCA_SUBBANDS], [SAMPLES_PER_SUBBAND]);
for (k = 0; k < s->audio_header.prim_channels; k++) {
int32_t (*subband_samples)[SAMPLES_PER_SUBBAND] =
s->dca_chan[k].subband_samples[block_index];
s->fmt_conv.int32_to_float(samples[0], subband_samples[0],
DCA_SUBBANDS * SAMPLES_PER_SUBBAND);
if (s->channel_order_tab[k] >= 0)
qmf_32_subbands(s, k, samples,
s->samples_chanptr[s->channel_order_tab[k]],
M_SQRT1_2 / 32768.0);
}
}
/* Generate LFE samples for this subsubframe FIXME!!! */
if (s->lfe) {
float *samples = s->samples_chanptr[ff_dca_lfe_index[s->amode]];
lfe_interpolation_fir(s,
s->lfe_data + 2 * s->lfe * (block_index + 4),
samples);
if (upsample) {
unsigned i;
/* Should apply the filter in Table 6-11 when upsampling. For
* now, just duplicate. */
for (i = 511; i > 0; i--) {
samples[2 * i] =
samples[2 * i + 1] = samples[i];
}
samples[1] = samples[0];
}
}
/* FIXME: This downmixing is probably broken with upsample.
* Probably totally broken also with XLL in general. */
/* Downmixing to Stereo */
if (s->audio_header.prim_channels + !!s->lfe > 2 &&
s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
dca_downmix(s->samples_chanptr, s->amode, !!s->lfe, s->downmix_coef,
s->channel_order_tab);
}
return 0;
}
static int dca_subframe_footer(DCAContext *s, int base_channel)
{
int in, out, aux_data_count, aux_data_end, reserved;
uint32_t nsyncaux;
/*
* Unpack optional information
*/
/* presumably optional information only appears in the core? */
if (!base_channel) {
if (s->timestamp)
skip_bits_long(&s->gb, 32);
if (s->aux_data) {
aux_data_count = get_bits(&s->gb, 6);
// align (32-bit)
skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
aux_data_end = 8 * aux_data_count + get_bits_count(&s->gb);
if ((nsyncaux = get_bits_long(&s->gb, 32)) != DCA_NSYNCAUX) {
av_log(s->avctx, AV_LOG_ERROR, "nSYNCAUX mismatch %#"PRIx32"\n",
nsyncaux);
return AVERROR_INVALIDDATA;
}
if (get_bits1(&s->gb)) { // bAUXTimeStampFlag
avpriv_request_sample(s->avctx,
"Auxiliary Decode Time Stamp Flag");
// align (4-bit)
skip_bits(&s->gb, (-get_bits_count(&s->gb)) & 4);
// 44 bits: nMSByte (8), nMarker (4), nLSByte (28), nMarker (4)
skip_bits_long(&s->gb, 44);
}
if ((s->core_downmix = get_bits1(&s->gb))) {
int am = get_bits(&s->gb, 3);
switch (am) {
case 0:
s->core_downmix_amode = DCA_MONO;
break;
case 1:
s->core_downmix_amode = DCA_STEREO;
break;
case 2:
s->core_downmix_amode = DCA_STEREO_TOTAL;
break;
case 3:
s->core_downmix_amode = DCA_3F;
break;
case 4:
s->core_downmix_amode = DCA_2F1R;
break;
case 5:
s->core_downmix_amode = DCA_2F2R;
break;
case 6:
s->core_downmix_amode = DCA_3F1R;
break;
default:
av_log(s->avctx, AV_LOG_ERROR,
"Invalid mode %d for embedded downmix coefficients\n",
am);
return AVERROR_INVALIDDATA;
}
for (out = 0; out < ff_dca_channels[s->core_downmix_amode]; out++) {
for (in = 0; in < s->audio_header.prim_channels + !!s->lfe; in++) {
uint16_t tmp = get_bits(&s->gb, 9);
if ((tmp & 0xFF) > 241) {
av_log(s->avctx, AV_LOG_ERROR,
"Invalid downmix coefficient code %"PRIu16"\n",
tmp);
return AVERROR_INVALIDDATA;
}
s->core_downmix_codes[in][out] = tmp;
}
}
}
align_get_bits(&s->gb); // byte align
skip_bits(&s->gb, 16); // nAUXCRC16
/*
* additional data (reserved, cf. ETSI TS 102 114 V1.4.1)
*
* Note: don't check for overreads, aux_data_count can't be trusted.
*/
if ((reserved = (aux_data_end - get_bits_count(&s->gb))) > 0) {
avpriv_request_sample(s->avctx,
"Core auxiliary data reserved content");
skip_bits_long(&s->gb, reserved);
}
}
if (s->crc_present && s->dynrange)
get_bits(&s->gb, 16);
}
return 0;
}
/**
* Decode a dca frame block
*
* @param s pointer to the DCAContext
*/
static int dca_decode_block(DCAContext *s, int base_channel, int block_index)
{
int ret;
/* Sanity check */
if (s->current_subframe >= s->audio_header.subframes) {
av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
s->current_subframe, s->audio_header.subframes);
return AVERROR_INVALIDDATA;
}
if (!s->current_subsubframe) {
/* Read subframe header */
if ((ret = dca_subframe_header(s, base_channel, block_index)))
return ret;
}
/* Read subsubframe */
if ((ret = dca_subsubframe(s, base_channel, block_index)))
return ret;
/* Update state */
s->current_subsubframe++;
if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {
s->current_subsubframe = 0;
s->current_subframe++;
}
if (s->current_subframe >= s->audio_header.subframes) {
/* Read subframe footer */
if ((ret = dca_subframe_footer(s, base_channel)))
return ret;
}
return 0;
}
static float dca_dmix_code(unsigned code)
{
int sign = (code >> 8) - 1;
code &= 0xff;
return ((ff_dca_dmixtable[code] ^ sign) - sign) * (1.0 / (1U << 15));
}
static int scan_for_extensions(AVCodecContext *avctx)
{
DCAContext *s = avctx->priv_data;
int core_ss_end, ret = 0;
core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;
/* only scan for extensions if ext_descr was unknown or indicated a
* supported XCh extension */
if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) {
/* if ext_descr was unknown, clear s->core_ext_mask so that the
* extensions scan can fill it up */
s->core_ext_mask = FFMAX(s->core_ext_mask, 0);
/* extensions start at 32-bit boundaries into bitstream */
skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
while (core_ss_end - get_bits_count(&s->gb) >= 32) {
uint32_t bits = get_bits_long(&s->gb, 32);
int i;
switch (bits) {
case DCA_SYNCWORD_XCH: {
int ext_amode, xch_fsize;
s->xch_base_channel = s->audio_header.prim_channels;
/* validate sync word using XCHFSIZE field */
xch_fsize = show_bits(&s->gb, 10);
if ((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&
(s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))
continue;
/* skip length-to-end-of-frame field for the moment */
skip_bits(&s->gb, 10);
s->core_ext_mask |= DCA_EXT_XCH;
/* extension amode(number of channels in extension) should be 1 */
/* AFAIK XCh is not used for more channels */
if ((ext_amode = get_bits(&s->gb, 4)) != 1) {
av_log(avctx, AV_LOG_ERROR,
"XCh extension amode %d not supported!\n",
ext_amode);
continue;
}
/* much like core primary audio coding header */
dca_parse_audio_coding_header(s, s->xch_base_channel);
for (i = 0; i < (s->sample_blocks / 8); i++)
if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {
av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");
continue;
}
s->xch_present = 1;
break;
}
case DCA_SYNCWORD_XXCH:
/* XXCh: extended channels */
/* usually found either in core or HD part in DTS-HD HRA streams,
* but not in DTS-ES which contains XCh extensions instead */
s->core_ext_mask |= DCA_EXT_XXCH;
break;
case 0x1d95f262: {
int fsize96 = show_bits(&s->gb, 12) + 1;
if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)
continue;
av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n",
get_bits_count(&s->gb));
skip_bits(&s->gb, 12);
av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);
av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));
s->core_ext_mask |= DCA_EXT_X96;
break;
}
}
skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
}
} else {
/* no supported extensions, skip the rest of the core substream */
skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));
}
if (s->core_ext_mask & DCA_EXT_X96)
s->profile = FF_PROFILE_DTS_96_24;
else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))
s->profile = FF_PROFILE_DTS_ES;
/* check for ExSS (HD part) */
if (s->dca_buffer_size - s->frame_size > 32 &&
get_bits_long(&s->gb, 32) == DCA_SYNCWORD_SUBSTREAM)
ff_dca_exss_parse_header(s);
return ret;
}
static int set_channel_layout(AVCodecContext *avctx, int channels, int num_core_channels)
{
DCAContext *s = avctx->priv_data;
int i;
if (s->amode < 16) {
avctx->channel_layout = dca_core_channel_layout[s->amode];
if (s->audio_header.prim_channels + !!s->lfe > 2 &&
avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
/*
* Neither the core's auxiliary data nor our default tables contain
* downmix coefficients for the additional channel coded in the XCh
* extension, so when we're doing a Stereo downmix, don't decode it.
*/
s->xch_disable = 1;
}
if (s->xch_present && !s->xch_disable) {
avctx->channel_layout |= AV_CH_BACK_CENTER;
if (s->lfe) {
avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
s->channel_order_tab = ff_dca_channel_reorder_lfe_xch[s->amode];
} else {
s->channel_order_tab = ff_dca_channel_reorder_nolfe_xch[s->amode];
}
} else {
channels = num_core_channels + !!s->lfe;
s->xch_present = 0; /* disable further xch processing */
if (s->lfe) {
avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
s->channel_order_tab = ff_dca_channel_reorder_lfe[s->amode];
} else
s->channel_order_tab = ff_dca_channel_reorder_nolfe[s->amode];
}
if (channels > !!s->lfe &&
s->channel_order_tab[channels - 1 - !!s->lfe] < 0)
return AVERROR_INVALIDDATA;
if (num_core_channels + !!s->lfe > 2 &&
avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
channels = 2;
s->output = s->audio_header.prim_channels == 2 ? s->amode : DCA_STEREO;
avctx->channel_layout = AV_CH_LAYOUT_STEREO;
/* Stereo downmix coefficients
*
* The decoder can only downmix to 2-channel, so we need to ensure
* embedded downmix coefficients are actually targeting 2-channel.
*/
if (s->core_downmix && (s->core_downmix_amode == DCA_STEREO ||
s->core_downmix_amode == DCA_STEREO_TOTAL)) {
for (i = 0; i < num_core_channels + !!s->lfe; i++) {
/* Range checked earlier */
s->downmix_coef[i][0] = dca_dmix_code(s->core_downmix_codes[i][0]);
s->downmix_coef[i][1] = dca_dmix_code(s->core_downmix_codes[i][1]);
}
s->output = s->core_downmix_amode;
} else {
int am = s->amode & DCA_CHANNEL_MASK;
if (am >= FF_ARRAY_ELEMS(ff_dca_default_coeffs)) {
av_log(s->avctx, AV_LOG_ERROR,
"Invalid channel mode %d\n", am);
return AVERROR_INVALIDDATA;
}
if (num_core_channels + !!s->lfe >
FF_ARRAY_ELEMS(ff_dca_default_coeffs[0])) {
avpriv_request_sample(s->avctx, "Downmixing %d channels",
s->audio_header.prim_channels + !!s->lfe);
return AVERROR_PATCHWELCOME;
}
for (i = 0; i < num_core_channels + !!s->lfe; i++) {
s->downmix_coef[i][0] = ff_dca_default_coeffs[am][i][0];
s->downmix_coef[i][1] = ff_dca_default_coeffs[am][i][1];
}
}
ff_dlog(s->avctx, "Stereo downmix coeffs:\n");
for (i = 0; i < num_core_channels + !!s->lfe; i++) {
ff_dlog(s->avctx, "L, input channel %d = %f\n", i,
s->downmix_coef[i][0]);
ff_dlog(s->avctx, "R, input channel %d = %f\n", i,
s->downmix_coef[i][1]);
}
ff_dlog(s->avctx, "\n");
}
} else {
av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode);
return AVERROR_INVALIDDATA;
}
return 0;
}
/**
* Main frame decoding function
* FIXME add arguments
*/
static int dca_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
AVFrame *frame = data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int lfe_samples;
int num_core_channels = 0;
int i, ret;
float **samples_flt;
DCAContext *s = avctx->priv_data;
int channels, full_channels;
int upsample = 0;
s->exss_ext_mask = 0;
s->xch_present = 0;
s->dca_buffer_size = ff_dca_convert_bitstream(buf, buf_size, s->dca_buffer,
DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);
if (s->dca_buffer_size == AVERROR_INVALIDDATA) {
av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
return AVERROR_INVALIDDATA;
}
if ((ret = dca_parse_frame_header(s)) < 0) {
// seems like the frame is corrupt, try with the next one
return ret;
}
// set AVCodec values with parsed data
avctx->sample_rate = s->sample_rate;
avctx->bit_rate = s->bit_rate;
s->profile = FF_PROFILE_DTS;
for (i = 0; i < (s->sample_blocks / SAMPLES_PER_SUBBAND); i++) {
if ((ret = dca_decode_block(s, 0, i))) {
av_log(avctx, AV_LOG_ERROR, "error decoding block\n");
return ret;
}
}
/* record number of core channels incase less than max channels are requested */
num_core_channels = s->audio_header.prim_channels;
if (s->ext_coding)
s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];
else
s->core_ext_mask = 0;
ret = scan_for_extensions(avctx);
avctx->profile = s->profile;
full_channels = channels = s->audio_header.prim_channels + !!s->lfe;
ret = set_channel_layout(avctx, channels, num_core_channels);
if (ret < 0)
return ret;
avctx->channels = channels;
/* get output buffer */
frame->nb_samples = 256 * (s->sample_blocks / SAMPLES_PER_SUBBAND);
if (s->exss_ext_mask & DCA_EXT_EXSS_XLL) {
int xll_nb_samples = s->xll_segments * s->xll_smpl_in_seg;
/* Check for invalid/unsupported conditions first */
if (s->xll_residual_channels > channels) {
av_log(s->avctx, AV_LOG_WARNING,
"DCA: too many residual channels (%d, core channels %d). Disabling XLL\n",
s->xll_residual_channels, channels);
s->exss_ext_mask &= ~DCA_EXT_EXSS_XLL;
} else if (xll_nb_samples != frame->nb_samples &&
2 * frame->nb_samples != xll_nb_samples) {
av_log(s->avctx, AV_LOG_WARNING,
"DCA: unsupported upsampling (%d XLL samples, %d core samples). Disabling XLL\n",
xll_nb_samples, frame->nb_samples);
s->exss_ext_mask &= ~DCA_EXT_EXSS_XLL;
} else {
if (2 * frame->nb_samples == xll_nb_samples) {
av_log(s->avctx, AV_LOG_INFO,
"XLL: upsampling core channels by a factor of 2\n");
upsample = 1;
frame->nb_samples = xll_nb_samples;
// FIXME: Is it good enough to copy from the first channel set?
avctx->sample_rate = s->xll_chsets[0].sampling_frequency;
}
/* If downmixing to stereo, don't decode additional channels.
* FIXME: Using the xch_disable flag for this doesn't seem right. */
if (!s->xch_disable)
avctx->channels += s->xll_channels - s->xll_residual_channels;
}
}
/* FIXME: This is an ugly hack, to just revert to the default
* layout if we have additional channels. Need to convert the XLL
* channel masks to libav channel_layout mask. */
if (av_get_channel_layout_nb_channels(avctx->channel_layout) != avctx->channels)
avctx->channel_layout = 0;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples_flt = (float **) frame->extended_data;
/* allocate buffer for extra channels if downmixing */
if (avctx->channels < full_channels) {
ret = av_samples_get_buffer_size(NULL, full_channels - channels,
frame->nb_samples,
avctx->sample_fmt, 0);
if (ret < 0)
return ret;
av_fast_malloc(&s->extra_channels_buffer,
&s->extra_channels_buffer_size, ret);
if (!s->extra_channels_buffer)
return AVERROR(ENOMEM);
ret = av_samples_fill_arrays((uint8_t **) s->extra_channels, NULL,
s->extra_channels_buffer,
full_channels - channels,
frame->nb_samples, avctx->sample_fmt, 0);
if (ret < 0)
return ret;
}
/* filter to get final output */
for (i = 0; i < (s->sample_blocks / SAMPLES_PER_SUBBAND); i++) {
int ch;
unsigned block = upsample ? 512 : 256;
for (ch = 0; ch < channels; ch++)
s->samples_chanptr[ch] = samples_flt[ch] + i * block;
for (; ch < full_channels; ch++)
s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * block;
dca_filter_channels(s, i, upsample);
/* If this was marked as a DTS-ES stream we need to subtract back- */
/* channel from SL & SR to remove matrixed back-channel signal */
if ((s->source_pcm_res & 1) && s->xch_present) {
float *back_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel]];
float *lt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 2]];
float *rt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 1]];
s->fdsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);
s->fdsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);
}
}
/* update lfe history */
lfe_samples = 2 * s->lfe * (s->sample_blocks / SAMPLES_PER_SUBBAND);
for (i = 0; i < 2 * s->lfe * 4; i++)
s->lfe_data[i] = s->lfe_data[i + lfe_samples];
if (s->exss_ext_mask & DCA_EXT_EXSS_XLL) {
ret = ff_dca_xll_decode_audio(s, frame);
if (ret < 0)
return ret;
}
/* AVMatrixEncoding
*
* DCA_STEREO_TOTAL (Lt/Rt) is equivalent to Dolby Surround */
ret = ff_side_data_update_matrix_encoding(frame,
(s->output & ~DCA_LFE) == DCA_STEREO_TOTAL ?
AV_MATRIX_ENCODING_DOLBY : AV_MATRIX_ENCODING_NONE);
if (ret < 0)
return ret;
*got_frame_ptr = 1;
return buf_size;
}
/**
* DCA initialization
*
* @param avctx pointer to the AVCodecContext
*/
static av_cold int dca_decode_init(AVCodecContext *avctx)
{
DCAContext *s = avctx->priv_data;
s->avctx = avctx;
dca_init_vlcs();
avpriv_float_dsp_init(&s->fdsp, avctx->flags & AV_CODEC_FLAG_BITEXACT);
ff_mdct_init(&s->imdct, 6, 1, 1.0);
ff_synth_filter_init(&s->synth);
ff_dcadsp_init(&s->dcadsp);
ff_fmt_convert_init(&s->fmt_conv, avctx);
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
/* allow downmixing to stereo */
if (avctx->channels > 2 &&
avctx->request_channel_layout == AV_CH_LAYOUT_STEREO)
avctx->channels = 2;
return 0;
}
static av_cold int dca_decode_end(AVCodecContext *avctx)
{
DCAContext *s = avctx->priv_data;
ff_mdct_end(&s->imdct);
av_freep(&s->extra_channels_buffer);
av_freep(&s->xll_sample_buf);
av_freep(&s->qmf64_table);
return 0;
}
static const AVOption options[] = {
{ "disable_xch", "disable decoding of the XCh extension", offsetof(DCAContext, xch_disable), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM },
{ "disable_xll", "disable decoding of the XLL extension", offsetof(DCAContext, xll_disable), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM },
{ NULL },
};
static const AVClass dca_decoder_class = {
.class_name = "DCA decoder",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
AVCodec ff_dca_decoder = {
.name = "dca",
.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_DTS,
.priv_data_size = sizeof(DCAContext),
.init = dca_decode_init,
.decode = dca_decode_frame,
.close = dca_decode_end,
.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.profiles = NULL_IF_CONFIG_SMALL(ff_dca_profiles),
.priv_class = &dca_decoder_class,
};
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